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2016-01-15Merge upstream SHA 04cb763Alex Luebs
* git merge 04cb763 * See all upstream changes since the previous merge in branch aosp/upstream-master: git diff cb3f9bd..04cb763 * Modify webrtc/.gitignore to keep *.mk files. * Removed old files from *.mk files: - thread.cc - thread_posix.cc * Add new files to *.mk files: - event_tracer.cc * Android relevant upstream changes: - Make Beamforming dynamically settable for Android platform builds - Remove additional channel constraints when Beamforming is enabled in AudioProcessing - Use an explicit identifier in Config Change-Id: I384a4e8f6982c31c5bc70eef521bb2d90800b9a4
2016-01-14Add tests for verifying transport feedback for audio and video.upstream-master.oldStefan Holmer
BUG=webrtc:5263 R=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1589523002 . Cr-Commit-Position: refs/heads/master@{#11255}
2016-01-14Eliminate defines in talk/kjellander
Replace LINUX, OSX and IOS defines with WEBRTC_ prefixed versions. Remove no longer used defines from talk/build/common.gypi due to previously migrated sources (into webrtc/p2p and webrtc/libjingle). When this is rolled into Chromium, we can also clean up the platform defines in https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp NOTRY=True BUG=webrtc:5420 TESTED=Ran all compile trybots with --clobber flag. TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1588453005 Cr-Commit-Position: refs/heads/master@{#11254}
2016-01-14Revert of Update with new default boringssl no-aes cipher suites. Re-enable ↵sprang
tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ ) Reason for revert: We're getting boringssl version conflicts. Reverting for now. Original issue's description: > Update with new default boringssl no-aes cipher suites. Re-enable tests. > > This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part). > > BUG=webrtc:5381 > R=davidben@webrtc.org, henrika@webrtc.org > > Committed: https://crrev.com/31c8d2eac5aec977f584ab0ae5a1d457d674f101 > Cr-Commit-Position: refs/heads/master@{#11250} TBR=davidben@webrtc.org,henrika@webrtc.org,torbjorng@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5381 Review URL: https://codereview.webrtc.org/1586183002 Cr-Commit-Position: refs/heads/master@{#11253}
2016-01-14Remove assert which was incorrectly added to TcpPort::OnSentPacket.Stefan Holmer
TBR=pthatcher@webrtc.org BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1588083002 . Cr-Commit-Position: refs/heads/master@{#11252}
2016-01-14Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.Stefan Holmer
Chromium reported errors when building libjingle_nacl due to some methods used virtual instead of override when they were overriding the base class. My guess is that when one method starts using override, all other in the same class must too. R=tommi@webrtc.org TBR=pthatcher@webtrc.org BUG=4173 Review URL: https://codereview.webrtc.org/1589563003 . Cr-Commit-Position: refs/heads/master@{#11251}
2016-01-14Update with new default boringssl no-aes cipher suites. Re-enable tests.Torbjorn Granlund
This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part). BUG=webrtc:5381 R=davidben@webrtc.org, henrika@webrtc.org Review URL: https://codereview.webrtc.org/1550773002 . Cr-Commit-Position: refs/heads/master@{#11250}
2016-01-14Revert of Connect TurnPort and TCPPort to ↵tommi
AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ ) Reason for revert: Broke Chrome: https://build.chromium.org/p/tryserver.chromium.linux/builders/linux_chromium_chromeos_compile_dbg_ng/builds/143025/steps/compile%20%28with%20patch%29/logs/stdio FAILED: cd ../../third_party/libjingle; python ../../native_client/build/build_nexe.py --root ../.. --product-dir ../../out/Debug/xyz --config-name Debug -t ../../native_client/toolchain/ --arch pnacl --build newlib_plib --name ../../out/Debug/gen/tc_pnacl_newlib/lib/libjingle_nacl.a --objdir ../../out/Debug/obj/third_party/libjingle/libjingle_nacl.gen/pnacl_newlib-pnacl/libjingle_nacl "--include-dirs=../../out/Debug/gen/tc_pnacl_newlib/include ../.. \"../../out/Debug/gen\" ./source ../ ../../native_client_sdk/src/libraries ../../native_client_sdk/src/libraries/nacl_io/include ../../native_client_sdk/src/libraries/third_party/newlib-extras ../expat/files/lib ../boringssl/src/include" "--compile_flags=-O2 -g -Wall -fdiagnostics-show-option -Werror -Wno-unused-function -Wno-char-subscripts -Wno-c++11-extensions -Wno-unnamed-type-template-args -Wno-extra-semi -Wno-unused-private-field -Wno-char-subscripts -Wno-unused-function \"-std=gnu++11\" " --gomadir /b/build/goma "--defines=\"__STDC_LIMIT_MACROS=1\" \"__STDC_FORMAT_MACROS=1\" \"_GNU_SOURCE=1\" \"_POSIX_C_SOURCE=199506\" \"_XOPEN_SOURCE=600\" \"DYNAMIC_ANNOTATIONS_ENABLED=1\" \"DYNAMIC_ANNOTATIONS_PREFIX=NACL_\" \"NACL_BUILD_ARCH=x86\" V8_DEPRECATION_WARNINGS \"CLD_VERSION=2\" \"_FILE_OFFSET_BITS=64\" CHROMIUM_BUILD \"CR_CLANG_REVISION=255169-1\" COMPONENT_BUILD UI_COMPOSITOR_IMAGE_TRANSPORT \"USE_AURA=1\" \"USE_ASH=1\" \"USE_PANGO=1\" \"USE_CAIRO=1\" \"USE_DEFAULT_RENDER_THEME=1\" \"USE_LIBJPEG_TURBO=1\" \"USE_X11=1\" \"IMAGE_LOADER_EXTENSION=1\" \"ENABLE_WEBRTC=1\" \"ENABLE_MEDIA_ROUTER=1\" USE_PROPRIETARY_CODECS ENABLE_PEPPER_CDMS ENABLE_CONFIGURATION_POLICY ENABLE_NOTIFICATIONS \"ENABLE_HIDPI=1\" \"ENABLE_TOPCHROME_MD=1\" USE_UDEV DONT_EMBED_BUILD_METADATA \"DCHECK_ALWAYS_ON=1\" FIELDTRIAL_TESTING_ENABLED \"ENABLE_TASK_MANAGER=1\" \"ENABLE_EXTENSIONS=1\" \"ENABLE_PDF=1\" \"ENABLE_PLUGINS=1\" \"ENABLE_SESSION_SERVICE=1\" \"ENABLE_THEMES=1\" \"ENABLE_AUTOFILL_DIALOG=1\" \"ENABLE_BACKGROUND=1\" \"ENABLE_PRINTING=1\" \"ENABLE_PRINT_PREVIEW=1\" \"ENABLE_SPELLCHECK=1\" \"ENABLE_CAPTIVE_PORTAL_DETECTION=1\" \"ENABLE_APP_LIST=1\" \"ENABLE_SUPERVISED_USERS=1\" \"ENABLE_MDNS=1\" \"ENABLE_SERVICE_DISCOVERY=1\" V8_USE_EXTERNAL_STARTUP_DATA FULL_SAFE_BROWSING SAFE_BROWSING_CSD SAFE_BROWSING_DB_LOCAL EXPAT_RELATIVE_PATH FEATURE_ENABLE_SSL GTEST_RELATIVE_PATH HAVE_OPENSSL_SSL_H NO_MAIN_THREAD_WRAPPING NO_SOUND_SYSTEM WEBRTC_POSIX SRTP_RELATIVE_PATH SSL_USE_OPENSSL USE_WEBRTC_DEV_BRANCH \"timezone=_timezone\" XML_STATIC \"USE_LIBPCI=1\" \"USE_OPENSSL=1\" \"USE_OPENSSL_CERTS=1\"" "--link_flags=-B../../out/Debug/gen/tc_pnacl_newlib/lib " "--source-list=../../out/gypfiles/third_party/libjingle/pnacl_newlib.libjingle_nacl.source_list.gypcmd" In file included from ../webrtc/p2p/base/tcpport.cc:67: ../webrtc/p2p/base/tcpport.h:50:23: error: 'CreateConnection' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override] virtual Connection* CreateConnection(const Candidate& address, ^ ../webrtc/p2p/base/portinterface.h:71:23: note: overridden virtual function is here virtual Connection* CreateConnection( ^ In file included from ../webrtc/p2p/base/tcpport.cc:67: ../webrtc/p2p/base/tcpport.h:53:16: error: 'PrepareAddress' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override] virtual void PrepareAddress(); ^ ../webrtc/p2p/base/portinterface.h:63:16: note: overridden virtual function is here virtual void PrepareAddress() = 0; ^ (etc) Original issue's description: > Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. > > To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it. > > BUG=4173 > R=pthatcher@webrtc.org > > Committed: https://crrev.com/7307952a5bf63311e5f9a2a90089a06177e42716 > Cr-Commit-Position: refs/heads/master@{#11247} TBR=pthatcher@webrtc.org,stefan@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=4173 Review URL: https://codereview.webrtc.org/1586063002 Cr-Commit-Position: refs/heads/master@{#11249}
2016-01-14Re-land: "Use an explicit identifier in Config"aluebs
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS. Original CL: https://codereview.webrtc.org/1538643004/ TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1589573004 Cr-Commit-Position: refs/heads/master@{#11248}
2016-01-14Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.Stefan Holmer
To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it. BUG=4173 R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1577873003 . Cr-Commit-Position: refs/heads/master@{#11247}
2016-01-14Add ramp-up tests for transport sequence number with and w/o audio.Stefan Holmer
Also add a perf metric tracking the average network latency. The audio stream test is disabled for now since audio isn't included in bitrate allocation. BUG=webrtc:5263 R=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1582833002 . Cr-Commit-Position: refs/heads/master@{#11244}
2016-01-14Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().Sergey Ulanov
Previosly ToSesnsetiveString() wasn't working witn some implementations of inet_ntop(). Rewrote it to avoid that dependency. BUG=chromium:577344 R=pthatcher@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1584793004 . Cr-Commit-Position: refs/heads/master@{#11242}
2016-01-13Revert of Storing raw audio sink for default audio track. (patchset #7 ↵deadbeef
id:120001 of https://codereview.chromium.org/1551813002/ ) Reason for revert: tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach. Original issue's description: > Storing raw audio sink for default audio track. > > BUG=webrtc:5250 > > Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99 > Cr-Commit-Position: refs/heads/master@{#11230} TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5250 Review URL: https://codereview.webrtc.org/1588693002 Cr-Commit-Position: refs/heads/master@{#11241}
2016-01-13Re-enable tests that failed under Linux_Msan.marpan
Fixed in latest libvpx roll. Keep EndToEndTest.TransportSeqNumOnAudioAndVideo disabled on Win_DrMemory for now as it seems to time-out/too slow. TBR=stefan@webrtc.org, kjellander@webrtc.org BUG=webrtc:5402 NOTRY=True Review URL: https://codereview.webrtc.org/1577313003 Cr-Commit-Position: refs/heads/master@{#11240}
2016-01-13Revert of Use an explicit identifier in Config (patchset #4 id:60001 of ↵tommi
https://codereview.webrtc.org/1538643004/ ) Reason for revert: Reverting due to problem with roll: /b/build/slave/linux/build/src/buildtools/linux64/gn gen //out/Release '--args=ffmpeg_branding="Chrome" proprietary_codecs=true is_debug=false is_component_build=false use_goma=true goma_dir="/b/build/goma" symbol_level=1 dcheck_always_on=true' --check --runtime-deps-list-file=/b/build/slave/linux/build/src/out/Release/runtime_deps -> returned 1 ERROR at //third_party/webrtc/BUILD.gn:245:18: Item not found configs -= [ "//build/config/clang:find_bad_constructs" ] ^----------------------------------------- You were trying to remove "//build/config/clang:find_bad_constructs" from the list but it wasn't there. GN gen failed: 1 step returned non-zero exit code: 1 @@@STEP_FAILURE@@@ Original issue's description: > Use an explicit identifier in Config > > This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS. > > Committed: https://crrev.com/25249d92d3cf105bcc7b684c8924ccdbc9afcb93 > Cr-Commit-Position: refs/heads/master@{#11231} TBR=henrik.lundin@webrtc.org,stefan@webrtc.org,tommi@chromium.org,aluebs@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1586563003 Cr-Commit-Position: refs/heads/master@{#11239}
2016-01-13Add build_protobuf variable.kjellander
This makes it possible to use protobuffers with an external protobuf library instead of the one that comes with the WebRTC code. NOTRY=True Review URL: https://codereview.webrtc.org/1589433002 Cr-Commit-Position: refs/heads/master@{#11236}
2016-01-13Clean the code for external denoiser.jackychen
BUG=webrtc:5255 Review URL: https://codereview.webrtc.org/1578373003 Cr-Commit-Position: refs/heads/master@{#11235}
2016-01-13[rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on ↵danilchap
the way Class renamed to indicated use of the rtcp::Empty class: it can only be used as container for other rtcp packets. All tests that use Append function moved from rtcp_packet_unittest, even if they did not use Empty class. This is because there is plan to make RtcpPacket class lighter by moving Append functionality to CompoundPacket class. BUG=webrtc:5260 R=åsapersson Review URL: https://codereview.webrtc.org/1582503002 Cr-Commit-Position: refs/heads/master@{#11234}
2016-01-13Fix capture ntp time issue introduced with r11187.Stefan Holmer
I think the problem was that I only introduced delay in one direction, and the estimation assumes that the RTT is evenly divided between the send direction and the receive direction, which was true for the old test. BUG=chromium:576246 R=pbos@webrtc.org Review URL: https://codereview.webrtc.org/1577853005 . Cr-Commit-Position: refs/heads/master@{#11233}
2016-01-13Use an explicit identifier in Configaluebs
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS. Review URL: https://codereview.webrtc.org/1538643004 Cr-Commit-Position: refs/heads/master@{#11231}
2016-01-13Storing raw audio sink for default audio track.deadbeef
BUG=webrtc:5250 Review URL: https://codereview.webrtc.org/1551813002 Cr-Commit-Position: refs/heads/master@{#11230}
2016-01-13Convert channel counts to size_t.Peter Kasting
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1316523002 . Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12[rtp_rtcp] rtcp::Sli packet moved into own file and got Parse functiondanilchap
BUG=webrtc:5260 R=åsapersson Review URL: https://codereview.webrtc.org/1551893002 Cr-Commit-Position: refs/heads/master@{#11228}
2016-01-12Make :rtc_base_approved a public dep of :rtc_base.jbroman
It looks to me like targets :rtc_base_approved is logically a subset of :rtc_base, and so any targets depending on :rtc_base expect to also get access to the headers in :rtc_base_approved. Thus I think it's appropriate for :rtc_base to have :rtc_base_approved in public_deps, so that `gn check` will permit this without clients having to explicitly depend on both. NOTRY=True Review URL: https://codereview.webrtc.org/1578833002 Cr-Commit-Position: refs/heads/master@{#11227}
2016-01-12NetEq: Fix a typo in a commentHenrik Lundin
TBR=minyue@webrtc.org NOTRY=true Review URL: https://codereview.webrtc.org/1578223003 . Cr-Commit-Position: refs/heads/master@{#11226}
2016-01-12Slap deprecation notices on Pass methodskwiberg
There's no reason not to use std::move instead now that we can use the C++11 standard library. BUG=webrtc:5373 Review URL: https://codereview.webrtc.org/1531013003 Cr-Commit-Position: refs/heads/master@{#11225}
2016-01-12Fix test bug introduced in r11101.Stefan Holmer
BUG=chromium:572995 R=pbos@webrtc.org Review URL: https://codereview.webrtc.org/1578223002 . Cr-Commit-Position: refs/heads/master@{#11224}
2016-01-12Remove unused enum RTPDirections.terelius
BUG= Review URL: https://codereview.webrtc.org/1582523002 Cr-Commit-Position: refs/heads/master@{#11221}
2016-01-12Wire-up BWE feedback for audio receive streams.Stefan Holmer
Also wires up receiving transport sequence numbers. BUG=webrtc:5263 R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1535963002 . Cr-Commit-Position: refs/heads/master@{#11220}
2016-01-12Add unit test for stand-alone denoiser and fixed some bugs.jackychen
The unit test will run the pure C denoiser and SSE2/NEON denoiser (based on the CPU detection) and compare the denoised frames to ensure the bit exact. TBR=tommi@webrtc.org BUG=webrtc:5255 Review URL: https://codereview.webrtc.org/1492053003 Cr-Commit-Position: refs/heads/master@{#11216}
2016-01-12Remove additional channel constraints when Beamforming is enabled in ↵aluebs
AudioProcessing The general constraints on number of channels for AudioProcessing is: num_in_channels == num_out_channels || num_out_channels == 1 When Beamforming is enabled and additional constraint was added forcing: num_out_channels == 1 This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo. Review URL: https://codereview.webrtc.org/1571013002 Cr-Commit-Position: refs/heads/master@{#11215}
2016-01-12Make Beamforming dynamically settable for Android platform buildsaluebs
Review URL: https://codereview.webrtc.org/1563493005 Cr-Commit-Position: refs/heads/master@{#11213}
2016-01-11clang-format audio_device/mac.andrew
NOTRY=true Review URL: https://codereview.webrtc.org/1570063003 Cr-Commit-Position: refs/heads/master@{#11212}
2016-01-11Change DTLS default from 1.0 to 1.2 for webrtc.Guo-wei Shieh
This changes for standalone webrtc applications. BUG= R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1548733002 . Cr-Commit-Position: refs/heads/master@{#11211}
2016-01-11Update API for Objective-C RTCMediaSource.Jon Hjelle
BUG= R=tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1538263002 . Patch from Jon Hjelle <hjon@andyet.net>. Cr-Commit-Position: refs/heads/master@{#11210}
2016-01-11Move Objective-C video renderers to webrtc/api/objc.Jon Hjelle
BUG= R=tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1542473003 . Patch from Jon Hjelle <hjon@andyet.net>. Cr-Commit-Position: refs/heads/master@{#11209}
2016-01-11Update API for Objective-C RTCMediaStreamTrack.Jon Hjelle
R=tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1527143002 . Patch from Jon Hjelle <hjon@andyet.net>. Cr-Commit-Position: refs/heads/master@{#11208}
2016-01-11Update API for Objective-C RTCStats.Jon Hjelle
BUG= R=tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1540113002 . Patch from Jon Hjelle <hjon@andyet.net>. Cr-Commit-Position: refs/heads/master@{#11207}
2016-01-11[rtp_rtcp] rtcp::Tmmbr moved into own filedanilchap
BUG=webrtc:5260 R=åsapersson Review URL: https://codereview.webrtc.org/1575023002 Cr-Commit-Position: refs/heads/master@{#11206}
2016-01-11H.264: Default flags and pulling in openh264 and ffmpeg.hbos
Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg). These will be used in a follow-up CL that adds an encoder and decoder implementation. The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added. BUG=468365 Review URL: https://codereview.webrtc.org/1575913003 Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11Move RTCI420Frame to webrtc/api/objc/RTCVideoFrame with minor style changes.Jon Hjelle
BUG= R=tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1533193003 . Patch from Jon Hjelle <hjon@andyet.net>. Cr-Commit-Position: refs/heads/master@{#11203}
2016-01-11[rtp_rtcp] rtcp::Tmmbn moved into own filedanilchap
explicetly unchanged. BUG=webrtc:5260 R=åsapersson Review URL: https://codereview.webrtc.org/1578713002 Cr-Commit-Position: refs/heads/master@{#11201}
2016-01-09fix GN build break on native_clientGuo-wei Shieh
TBR=guidou@webrtc.org BUG= Review URL: https://codereview.webrtc.org/1576723002 . Cr-Commit-Position: refs/heads/master@{#11196}
2016-01-08Properly handle different transports having different SSL roles.Taylor Brandstetter
This meant splitting "transport_options" into audio/video/data options, for when creating the answer, and giving "GetSslRole" a "transport_name" parameter so we can retrieve the current role on a per-transport basis. BUG=webrtc:4525 R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1516993002 . Cr-Commit-Position: refs/heads/master@{#11192}
2016-01-08Misc. small cleanups.pkasting
* Better param names * Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases. * Use arraysize() * Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers * reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead * Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition * Fix indenting * Use uint32_t for timestamps (matching how it's already a uint32_t in most places) * Spelling * RTC_CHECK_EQ(expected, actual) * Rewrap * Use .empty() * Be more pedantic about matching int/int32_t/ * Remove pointless consts on input parameters to functions * Add missing sanity checks All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first. BUG=none TEST=none Review URL: https://codereview.webrtc.org/1534193008 Cr-Commit-Position: refs/heads/master@{#11191}
2016-01-08Cleaning neteq_unittest resource files.minyue
BUG=webrtc:2692 Review URL: https://codereview.webrtc.org/1563983003 Cr-Commit-Position: refs/heads/master@{#11189}
2016-01-08Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mackjellander
NOTRY=True BUG=5407 TBR=stefan@webrtc.org,pbos@webrtc.org Review URL: https://codereview.webrtc.org/1569273003 Cr-Commit-Position: refs/heads/master@{#11188}
2016-01-08Add CreateSend/ReceiveTransport() methods to CallTest.stefan
This allows the test to create its own transports if it, for instance, needs to do demuxing. BUG=webrtc:5416 Review URL: https://codereview.webrtc.org/1573453002 Cr-Commit-Position: refs/heads/master@{#11187}
2016-01-08Reland "Add APK targets to build libjingle tests for Android."phoglund
patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ This reverts commit bc14164aad254e72ce4d1e381b912b7d3acf5391. We have made more preparations downstream, so this should work now. Original CL by perkj@. BUG=webrtc:2365 The work started from the work by kjellander@ in https://codereview.webrtc.org/1413663003/ Review URL: https://codereview.webrtc.org/1570513004 Cr-Commit-Position: refs/heads/master@{#11186}
2016-01-08Update attributes to match gclibc's ansidecl.hkjellander
To ease use of WebRTC in other codebases, update some macros to match glibc's ansidecl.h, which uses double-underscores for attributes. NOTRY=True Review URL: https://codereview.webrtc.org/1571653002 Cr-Commit-Position: refs/heads/master@{#11185}
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